similar to: asterisk queue - skills based routing (patch updated)

Dne 6.8.2015 v 21:00 Sylvain Boily napsal(a): > Hello, > > Le 2015-08-06 09:24, Marek Cervenka a ?crit : >> hi, >> >> there is updated skills based routing patch for asterisk queue >> please test if you have time >> >> https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22 >> >> > > You can find the latest

Hi, Le ven. 10 juil. 2020 à 16:56, Sylvain Boily <sylvain at wazo.io> a écrit : > > It probably can help you: > > https://github.com/wazo-platform/wazo-provd-plugins/blob/master/plugins/xivo-jitsi/1/templates/base.tpl > > Sylvain > > Yes, provided example was exactly what I was after ! Thank you very much ! -------------- next part -------------- An HTML attachment

Hello, I did some tests because i'm interesting to transfer a non stasis bridge to a stasis bridge and i found a strange situation. A call B B answer You have a bridge On my asterisk CLI: xivo*CLI> bridge show b1d8fb21-ec6d-469a-9dde-bb6bfd5618cc Id: b1d8fb21-ec6d-469a-9dde-bb6bfd5618cc Type: basic Technology: simple_bridge Num-Channels: 2 Channel: SIP/tcu9tz-00000032 Channel:

Hello, 1. I'm looking for an (anomized) example of a Jitsi Desktop provisioning file compliant with Asterisk ? Jitsi Doc mentions it should adhere to Java properties file syntax (see [1]) but a working example would help. If this example file included the following settings, it would be perfect: - sip username, secret, authname, ... - codec G711, RFC2833 - BLF - local or remote phonebook -

Hi, I want to pass a part of a SIP INVITE multipart body. I found a quite old patch here: https://issues.asterisk.org/jira/browse/ASTERISK-14510?jql=text%20~%20%22body%20part%22 But this patch is for the SIP channel driver not PJSIP, right? Is it even possible without a patch? What do I have to put in the dialplan then? Thanks in advance, Simon

Le 2015-12-15 15:25, Joshua Colp a ?crit : > Sylvain Boily wrote: >> Hello, > > Just a note - there's an asterisk-app-dev mailing list[1] which is > better suited for these kind of posts. Ok > >> I did some tests because i'm interesting to transfer a non stasis bridge >> to a stasis bridge and i found a strange situation. > > You can't, you have

hi, i''m trying move from file: based disk to tap:aio but things don''t work i have centos4 dom0 with centos4 domU xen 3.0.4-testing changeset: 13138:d401cb96d8a0 self compiled [root@xen linux-2.6.16.38-xen]# grep XEN_BLKDEV_TAP .config CONFIG_XEN_BLKDEV_TAP=m config disk = [ ''file:/var/lib/xen/test.img,hda1,w'',

Hi all, For anyone that's interested in setting up TLS with libvirt, but hasn't gotten around to it yet, there's a decent guide on the wiki now: http://wiki.libvirt.org/page/TLSSetup It's user focused (ie SysAdmin's), and has lots of pics explaining concepts, plus shows what needs to be done, and shows how to do it. Please note, it doesn't cover SASL, it's just

hi all, Can someone help me with a kind of howto build call center around asterisk with all the necessary features like CTI, call recordings, call spying, real time monitoring etc? I will be glad if it is an open source code. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello all, I am looking for an implementation of a 10 man call center. low cost license or GPL will be preferred. I will be glad for your help. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160622/8e2e5ab8/attachment.html>

hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? thanks -- --------------------------------------- Marek

hi, i have the same problems as in https://issues.asterisk.org/jira/browse/ASTERISK-25833 my current combination is centos 6 32-bit, unixODBC 2.3.2 (recompiled from fedora20), mysql 5.1.73, mysql-connector-odbc 5.1.5, asterisk 13.9.1 i tried update to mysql-connector-odbc 5.3.6 from oracle but it segfaults every time can you share your working mysql+odbc combination? thanks odbcinst.ini

hi, after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i have problem with segfault (centos 6) Program terminated with signal 11, Segmentation fault. #0 0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8, pkt_size=132, options=1, token=0xafecc2bc, parsed_len=0x0, src_addr=0xb0e47a20, src_addr_len=16) at ../src/pjnath/stun_session.c:1287 1287

please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compile&run clients with srtp (linksys,gxp-2000, minisip, twikle, ...) --------------------------------------- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA - http://lcna.slu.cz =======================================

hi, can you recommend "clean&simple&stable" solution for incoming call popup (in browser)? i'm using flash operator panel now but i want something without flash (maybe something in AJAX?) thanks --------------------------------------- Marek Cervenka =======================================

hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is

doesnt work for me Dne 29.5.2016 v 17:48 Niklas Larsson napsal(a): > Hi, > > > On 2016-05-27 18:28, Marek ?ervenka wrote: >> after downgrade to 13.8.2 >> May 27 18:21:06 ast kernel: asterisk[16286]: segfault at 1010024 ip >> b49162cd sp bfac0940 error 4 in >> libmysqlclient.so.16.0.0[b48f1000+12e000] >> >> after downgrade to 13.7.2 >>

The time appears to be perfect inside dom0, however all the domU''s tend to have a slightly faster date which gets further out of sync every day. I''m currently using Xen 3.0.3 with Gentoo Linux, under 3.0.2 I had no problems with domU clocks. Are there any known issues which could cause this? I''d strongly prefer not to run ntpd in every domU, having all domU clocks in

hi, do you have someone example of http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/ in node.js asterisk-ari ? thanks Marek

On Thu, Oct 20, 2016 at 4:50 AM, marek cervenka <cervajs2 at gmail.com> wrote: > i tested this > > # cat /etc/asterisk/extconfig.conf > [settings] > queue_log => sqlite3,cdrDb > > # cat /etc/asterisk/res_config_sqlite3.conf > [cdrDb] > dbfile = /var/lib/asterisk/realtime.sqlite3 > > sqlite3 /var/lib/asterisk/realtime.sqlite3 > > CREATE TABLE